Call landing issues in NAT

1. Brekeke Product Name and Version:

2. Java version: 1.5.0_06

3. OS type and the version: Linux 2.6.9-89.ELsmp

4. UA (phone), gateway or other hardware/software involved:

5. Your problem:

i have two sip client installed in my mobile.both are behind NAT.
i have configured Public IP address in sip server.both client registering successfully with sip server.my problem is some time call works well.some time INVITE message not received called party.so the call is in initiating state. is i am missing to configure anything.?

i have wireshark trace for working and not working state. if need i can send.and also let me know where the option to upload wireshark trace in this forum.

Regards
Bens

There are several ways to solve the issue.

First, set a shorter value at the [Interval (ms)] field in the [Configuration]->[SIP] page.


Let me know if the issue still happens.

Hi laurie.
Thanks for your suggestion. i will check the setting as you said.and let you know the status shortly.

Regards
Bensly

Hi,
Thanks for the solutions and its worked for me.but now i am facing another problem is both side no audio.but call successfully established. i am able to see the RTP packet goes to sip server from two end points.but sip server not delivering the RTP to the clients.kindly let me know how to solve this. in interface i have added public IP address.

there is no rtp packets sent from Brekeke server IP both public or private?
which version sip server are you using?
try with latest v3.1.7.8

Hi,
i have upgraded to v3.1.7.8. but still the issue there.i am creating new post for this.please replay on the new post.

Regards
Bensly